Dynamic volume control and multi-spatial processing protection

ABSTRACT

A disclosed system and method dynamically controls the perceived volume of a stereo audio program including left and right channel signals. The system comprises: a dynamic volume control configured and arranged so as to maintain a perceived constant volume level of the stereo audio program; and an excessive spatial processing protection processor configured and arranged for controlling the level of a difference signal, created as a function of the right channel signal subtracted from the left channel signal (L−R), relative to the level of a sum signal, created as a function of the right channel signal plus the left channel signal;; wherein the excessive spatial processing protection processor processes the audio signals so as to control the difference (L−R) signal relative to the sum (L+R) signal. A system and method are also provided for dynamically controlling the perceived volume of a stereo audio program including left and right channel signals, comprising: a dynamic volume control configured and arranged so as to maintain a perceived constant volume level of the stereo audio program; and a program change detector configured and arranged to provide a program change signal indicating that the volume of the left and right channel signals has dropped below a threshold level for at least a threshold time period so as to anticipate a possible change in the sound level of the left and right channel signals; wherein the dynamic volume control is responsive to the program change signal.

RELATED APPLICATIONS

This application is related to and claims priority to U.S. ProvisionalApplication Nos. 61/114,684 filed on 14 Nov. 2008 in the names ofChristopher M. Hanna, Gregory Benulis and Scott Skinner; and 61/114,777filed on 14 Nov. 2008 in the names of Christopher M Hanna and GregoryBenulis, both applications being herein incorporated by reference. Thisapplication is also related to copending U.S. application Ser. No.______ (Attorney's Docket No. 56233-428-THAT-27) contemporaneously filedwith the present application in the names of, Christopher M. Hanna andGregory Benulis, and assigned to the present assignee.

TECHNICAL FIELD

The present application relates to audio signal processing, and moreparticularly to audio signal volume control and multi-spatial processingprotection.

BACKGROUND

During television viewing, volume changes can be irritating and ofteninvolve manual volume adjustments by the viewer. One example is theperceived volume change that often occurs when changing channels on atelevision. Another example would be the perceived volume change thatcan occur between the broadcast of a television program and acommercial. These large relative changes are typically attributed tolack of level control at the point of broadcast or signal compressionintroduced during production. A somewhat little known cause of increasedperceived loudness is multiple spatial processing. The audio in someprogram material is processed, in the studio, to introduce surroundspatial effects (pseudo-surround) in two-channel systems. If this typeof broadcast audio is then processed in the television to introducetwo-channel surround effects, as is currently done in many televisionmodels, the perceived level change can be dramatic. This additionalspatial processing can cause the center image (typically dialogue) to bealmost unintelligible. In all cases automatic volume control technologycan minimize listener discomfort and maintain a more consistent volumelevel. While much attention has been paid to leveling the audio volumeat the point of broadcast, it seems to have done little to alleviate theproblem. In fact, with the advent of high dynamic range DTV broadcastswider loudness differences can be now perceived by the televisionviewer.

SUMMARY

In accordance with one aspect of the disclosed system and method, asystem is provided for dynamically controlling the perceived volume of astereo audio program including left and right channel signals,comprising: a dynamic volume control configured and arranged so as tomaintain a perceived constant volume level of the stereo audio program;and an excessive spatial processing protection processor configured andarranged for controlling the level of a difference signal created as afunction of the right channel signal subtracted from the left channelsignal (L−R) relative to the level of a sum signal created as a functionof the right channel signal plus the left channel signal; wherein theexcessive spatial processing protection processor processes the audiosignals so as to control the difference (L−R) signal enhancement.

In accordance with another aspect, a system is provided for dynamicallycontrolling the perceived volume of a stereo audio program includingleft and right channel signals, comprising: a dynamic volume controlconfigured and arranged so as to maintain a perceived constant volumelevel of the stereo audio program; and a program change detectorconfigured and arranged to provide a program change signal indicatingthat the volume of the left and right channel signals has dropped belowa threshold level for at least a threshold time period so as toanticipate a possible change in the sound level of the left and rightchannel signals; wherein the dynamic volume control is responsive to theprogram change signal.

In accordance with yet another aspect, a system is provided fordynamically controlling the perceived volume of a stereo audio programincluding left and right channel signals, comprising: a dynamic volumecontrol configured and arranged so as to maintain a perceived constantvolume level of the stereo audio program, the dynamic volume controlincluding at least compressor responsive to high and low attack andrelease ratio thresholds so as define quiet, normal and loud perceivedvolume levels.

In accordance with still another aspect, a system is provided fordynamically controlling the perceived volume of a stereo audio programincluding left and right channel signals, comprising: an excessivespatial processing protection processor configured and arranged forcontrolling the level of a difference signal created from subtractingthe right channel signal from the left channel signal (L−R), and acontour filter for shaping the difference signal.

In accordance with yet another aspect, a system is provided fordynamically controlling the perceived volume of a stereo audio programincluding left and right channel signals. The system comprises: anexcessive spatial processing protection processor configured andarranged for controlling the level of a difference signal created fromsubtracting the right channel signal from the left channel signal (L−R),and a contour filter for shaping the difference signal.

GENERAL DESCRIPTION OF THE DRAWINGS

The drawings disclose illustrative embodiments. They do not set forthall embodiments. Other embodiments may be used in addition or instead.Details that may be apparent or unnecessary may be omitted to save spaceor for more effective illustration. Conversely, some embodiments may bepracticed without all of the details that are disclosed. When the samenumeral appears in different drawings, it refers to the same or likecomponents or steps.

Aspects of the disclosure may be more fully understood from thefollowing description when read together with the accompanying drawings,which are to be regarded as illustrative in nature, and not as limiting.The drawings are not necessarily to scale, emphasis instead being placedon the principles of the disclosure. In the drawings:

FIG. 1 is a simplified block diagram of one embodiment of a dynamicvolume control system;

FIG. 2 is a state diagram illustrating one embodiment of the operationof one program change detection;

FIG. 3 is a simplified block diagram of one embodiment of a single bandof a dynamic volume control system;

FIG. 4 is a simplified block diagram of one embodiment of a multi-banddynamic volume control system;

FIGS. 5-7 graphically illustrate frequency responses of a multi-banddynamic volume control system;

FIG. 8 is a simplified block diagram of one embodiment of a doubleprocession protection system;

FIG. 9 is a simplified block diagram of one embodiment of an arrangementof a combined system including both a dynamic volume control system anda double processing protection system; and

FIG. 10 is a simplified block diagram of a second embodiment of anarrangement of a combined system including both a dynamic volume controlsystem and a double processing protection system

DETAILED DESCRIPTION OF THE DRAWINGS

Illustrative embodiments are now discussed. Other embodiments may beused in addition or instead. Details that may be apparent or unnecessarymay be omitted to save space or for a more effective presentation.Conversely, some embodiments may be practiced without all of the detailsthat are disclosed.

Dynamic Volume Control (DVC) System

A DVC system is described for dynamically controlling the volume of anaudio signal. The system is configured and arranged so as to dynamicallymanipulate and modify sound volume when sudden changes occur. Theembodiments described herein are configured and arranged so as tomaintain a perceived constant volume level for audio band applications.The DVC system can be entirely digital and can be implementedeconomically in software (C, assembler etc.) or digital hardware (HDLdescription), although it should be evident that the system can beentirely analog, or a hybrid analog/digital system. Market applicationsinclude television audio, DVD player audio, set top box audio, radioaudio and other hifi and non-hifi audio products. Without a DVC systemof the type described herein, perceived volume levels can varydramatically as program material changes within a given broadcast/sourceor as the audio broadcast/source changes. These volume changes can beirritating and often involve manual volume adjustments by the listener.One specific example would be the volume changes that occur whenchanging channels on a television. Another example would be the volumechanges between a television program and a television commercial. Inboth examples the DVC system would eliminate listener discomfort andmaintain a more consistent volume level.

FIG. 1 shows one embodiment of such a DVC system 100. The system 100receives two input signals, a left signal L at input 102 and a rightsignal at input 104. In the embodiments described the DVC systemarchitecture is based upon a digital implementation of a classiccompressor design (THAT Corporation Design Note 118) with flexibilityand additional modifications that are only possible in a digitalimplementation. System 100 includes an RMS level detector 110 forproviding a signal representative of the sum of the RMS averages of theleft and right signals L and R, log conversion block 112, and a signalaveraging AVG block 114. Log conversion block 112 converts the output ofthe RMS level detector 110 from the linear to the logarithmic domain.System 100 is responsive to a number of control signals each indicativeof whether a certain condition exists requiring a response from thesystem The system 100 also includes a host processor (not shown)configured and arranged for carrying out the operating of the DVC system100. The illustrated embodiment is responsive to a number of controlsignals including: a target level signal provided by the target signalgenerating device 116, an attack threshold signal generated by theattack threshold signal device 118, a release threshold (not shown), agate threshold signal generated by the gate threshold signal device 120,an attack ratio threshold (not shown), a release ratio threshold (notshown), a ratio signal generated by the ratio signal device 122, and amute hold signal generated by mute hold device 124 responsive to aprogram change detector (PCD-not shown). Devices 116, 118, 120, 122 cansimply be adjustable user controls accessible to the user. Device 124can be arranged to receive a signal from the TV controls when thechannel changes or from a mute detector (not shown) that detects ifinputs 102 and 104 have both been muted. The target signal level 116represents the level in dB, relative to a full scale input, that is thetarget volume. The attack threshold 118 represents the number of dB thatREF must be above AVG before the attack time is reduced by a factor ofN, where N can be any number. In one illustrated embodiment N=10. Therelease threshold signal represents the number of dB that REF must bebelow AVG before the release time is reduced by a factor of M, where Mcan be any number, and in one illustrated embodiment M=10 The Gatethreshold 120 represents the amount, a negative dB number, that REF cango below AVG before all left and right gain adjustments are frozen. Theattack ratio threshold represents the absolute amount, in dB, that REFcan go above the target signal level 116 before the volume controlbegins attenuating the input signal. The release ratio thresholdrepresents the absolute amount, in dB, that REF can go below the targetsignal level 116 before the volume control begins adding gain to theinput signal. The ratio signal 122 adjusts the AVG value by the desiredcompression ratio.

Target level signal 116 is subtracted from the output of log conversionblock 112 by signal summer 126 so as to provide the REF signal to thesignal averaging AVG block 114, a comparator 128 and a second comparator130. The REF signal represents the volume level of the input signalrelative to the desired listening threshold. The AVG signal can also bethought of as the instantaneous (prior to attack/release processing)ideal gain recommendation. The output of the signal averaging block 114is the AVG signal, which is a signal that is a function of the averageof the REF signal. The AVG signal is applied to the signal summer 132where it is added to the attack threshold signal 118. In a similarmanner (not shown) the AVG signal is summed with a release threshold.The AVG signal is also applied to the signal summer 134 where it isadded to the gate threshold signal 120. The output of signal summer 132is applied to attack threshold comparator 128 where it is compared tothe REF signal, while the output of signal summer 134 is applied to gatethreshold comparator 130 where it is compared to the REF signal. The AVGsignal is also multiplied by the ratio signal 122 by the signalmultiplier 136. The output of comparator 128 is applied to theattack/release selection block 138, which in turn provides either an Att(attack) signal, or a Rel (release) signal to the signal averaging block114, dependent on and responsive to the status of the mute hold signal124. The output of the release threshold AVG summer (not shown) is alsocompared to the REF signal and is applied to the attack/releaseselection block. The comparator 130 provides an output to the HOLD inputof signal averaging block 114. Finally, the signal multiplier 136provides an output to a log-to-linear signal converter 140, which inturn provides an output which is applied to each of the signalmultipliers 142 and 144, wherein it respectively scales the left andright signal provided at the corresponding inputs 102 and 104 so as toprovide the output modified left and right signals Lo and Ro.

Referring to FIG. 1, the RMS level detector 110 senses the sound levelof the input signal. It should be noted that while an RMS level detectoris shown, any type of signal level detector can be used. For example, apeak detector, average detector, perception based level detector (suchas the ITU 1770 loudness detector or the CBS loudness detector), orother detector can be used to sense the sound level. These leveldetectors usually have time constants which are dynamically andindependently adjustable. One method of adjusting these time constantsis to base them on the envelope or general shape of the input signal sothat the time constants vary with the signal. In other embodiments, thetime constants are fixed. For ease of data processing, the sound levelcan be converted into the log domain, as shown, using log conversionblock 112. In a multi-band system, a separate RMS detector can be usedfor each band. The signal averaging block 114 is configured and arrangedso as to compute the average of REF relative to the attack and releasetimes. The output signal AVG of the signal averaging block 114 isadjusted by the desired compression ratio, via multiplier 136, to createthe gain value to be applied. Finally the gain is converted back intothe linear domain by the log-to-linear converter 140 for application tothe left and right signals L and R so as to produce the modified leftand right signals Lo and Ro.

A target output level represented by the target level signal 116 issubtracted from the sensed level at the output of the log conversionblock 112 to determine the difference between the actual and desiredsound level. This difference, which represents the level of the inputsignal relative to the target level signal 116, is known as thereference (REF) signal. The target level signal can be a user input,such as a simple knob or other pre-set setting, so as to control thelevel of sound desired. This threshold can be fixed or it can be changedas a function of the input signal level to better position thecompression relative to the input dynamic range. Once REF signal isobtained, it is provided as an input to the averaging block 114, attackthreshold comparator 128 and gate threshold comparator 130. The outputof attack threshold comparator 128 is applied to the attack/releaseselect block 138, which in turn receives a signal a MuteHold signals 124from a program change detector.

The gate threshold signal 120 when added to the current average AVGrepresents the lowest value REF is able to achieve before left and rightgain adjustment (142 and 144) are frozen. The gate threshold comparator130 receives the instantaneous signal level (REF) signal and determinesif the sound level represented by REF drops below the givenaforementioned threshold. If the instantaneous signal level (REF) ismore than the amount of the gate threshold below the averaged signallevel (AVG) appearing at the output of block 114, the gain applied tothe signal in the signal path is held constant until the signal levelrises above the threshold. The intent is to keep the system 100 fromapplying increased gain to very low level input signals such as noise.In an infinite hold system, the gain can be constant forever until thesignal level rises. In a leaky hold system, the gain can be increased ata gradual pace (much slower than the release time). In a one embodiment,this gate hold threshold is adjustable, while in another embodiment thethreshold set by gate threshold 134 is fixed.

The program change detector, or mute-hold, senses when the input is“silent.” When a user changes a television (TV) channel, the sound levelbetween the two channels can change, either increasing or decreasingsignificantly. Typically, a television manufacturer will mute the audiobriefly while changing channels to protect the viewer from irritatingaudio transients. The program change detector is designed to check forthis sort of muting by determining if the sound level drops below apredetermined threshold (MuteLev) for a predetermined amount of time(MuteTime). If the instantaneous sound level (REF) is below thethreshold for a certain period of time, or “mute time,” then a programchange is detected. If a program change is detected the speeds of theattack and release times (described in further detail below) areincreased. With this increase, if a loud channel is changed to a quietchannel, then the increased release time permits a faster gain increaseto meet the target sound output level. Conversely, if a quiet channel ischanged to a loud channel, then the increased attack time permits afaster gain decrease to meet the target. If the sound level rises abovethe threshold before the “mute time” expires, then a program change isnot detected. In alternative embodiments, the “mute time” and the mutethreshold can be fixed, user adjustable, variable, or otherwise.

FIG. 2, illustrates one embodiment of state diagram of a mute detectionalgorithm for the operation of the program change detector. Theoperation 200 includes three states, the MUTE OFF state 202, the MUTE ONstate 208 and the MUTE HOLD state 212. In the MUTE OFF state 202 the REFsignal at the output of the signal summer 126 is periodically comparedto MuteLev threshold level at 204 to determine whether REF>MuteLev orREF<MuteLev. If REF>MuteLev, then the operation remains in state 202 andcontinues in that state. In this state, MUTE ON=0, MUTE HOLD=0, and theattack and release times are at their normal settings. If, however,Ref<MuteLev, a mute is detected and the operation transitions at 206 tostate 208 MUTE ON. Once transitioned to state 208, MUTE ON=1, and in thestate 208, the program change detector next determines whether the mutecondition remains for a predetermined time. If the condition of Mutedoes not last long enough and REF>MuteOffLev occurs before theexpiration of the timer, the detector transitions back to the state 202.This might occur where there is a pause in program where the audioportion is silent. However, where the timer determines that the MuteTime has been expired, a program change has occurred. In this state whenthe REF>MuteOffLev returns, the detector will transition at 210 to theMUTE HOLD state 212. In this state, the attack and release times aresped up so that a relative loud signal is made softer, and a relativelysoft signal is made louder for a predetermined time limit (Mute Time).In FIG. 2 the timer setting in state 208 is shown to be the same as instate 212. It should be obvious that they can also be different values.While in state 212, if the Ref decreases below the MuteLev setting(i.e., Ref<MuteLev) prior to the expiration of the Mute Time, the statetransitions at 214 back to state 208. If, however, the Mute Time doesexpire the detector will transition at 216 back to the state 202.

In one implementation the MuteTime and MuteLev (mute level) areadjustable. The mute time and mute level can also be fixed in a givenimplementation. The mute threshold is set lower than the gate threshold.The mute detection algorithm can function in an automatic or manualmode. In automatic mode the system 100 detects the mute condition duringa channel change. The program change detector can also operate in amanual mode, where a “muting” signal is received from a television orother device indicating that a channel is being changed. Further, theprogram change detector can also receive signals from a user's remotecontrol to interpret whether the user is changing a channel. The system100 can also operate using attack and release thresholds. If, in a giventime window, a sound level jumps to the extent that the attack threshold118 is traversed, then the system 100 can operate in “fast attack” mode.In one embodiment, if REF exceeds AVG by the attack threshold, this fastattack mode increases the attack time constant to quickly reduce thegain of this increased sound level. Similarly, if the release thresholdis traversed, then the system operates in fast release mode, where thegain is increased quickly. These attack and release time constants canbe independently adjustable between each other and also between high andlow bands in a multi-band system.

In some implementations the maximum gain applied to the input signal maybe limited. This would limit the amount of gain applied to a quiet audiopassage. If a loud passage (thunder in a movie) immediately followed thequiet audio passage, unlimited gain could result in significant audioovershoot before the gain could be reduced over the attack time.

Averaging block 114 receives the REF, attack, release and hold signalsand determines the average (AVG) of the REF signal based on and as afunction of the attack, release, and hold signals. The AVG signal isthen adjusted by the compression ratio to be applied to the originalsignal for volume control. The AVG signal represents the REF signalprocessed with the Attack/Release time constants. Once a change in REFripples through the averaging block 114 to affect the AVG signal, itfirst needs to be adjusted by the desired compression ratio. It shouldbe appreciated that system 100 does not compress infinitely. Once thevalue of the AVG signal is adjusted by the compression ratio, the AVGsignal is multiplied by −(1-ratio) via ratio setting device 122 andmultiplier 136. Thus, by way of example, a 4:1 compression ratio wouldmultiply the AVG signal by −(1−1/4) or −3/4. So if the audio is 20 dBabove the threshold value, the AVG signal would equal 20 dB (after theattack time constant has elapsed). Multiplying 20 dB by −3/4 yields avalue of −15 dB. As a result the audio that is 20 dB over the thresholdis attenuated to 5 dB after the −15 dB gain is applied. 20/5=4 which isa 4:1 compression ratio.

The compression ratio applied to the signal can be a single slopedratio. For example, a 4:1 ratio can be applied to the incoming signal,depending on the level threshold. If AVG is above the threshold, thenthe signal would be reduced by a factor of four (at the attack rate).Conversely, if AVG is below the threshold, then the signal would beamplified by a factor of four (at the release rate).

In another embodiment, the compression ratio can be different, dependingon whether the AVG signal is above or below the Target Level thresholdprovided by device 116. For example, if the AVG signal is above theTarget Level threshold, then the signal can be reduced by a factor offour, as in the previous example. In contrast, however, if AVG is belowthe threshold, then a different ratio can be applied to amplify theinput signal, say a 1.5:1 ratio. This arrangement permits thecompression of loud signals above the ratio threshold, but alsopreserves the sound level for quiet dialogue, such as whispers. Thearrangement described above could be thought of as a movie mode; ittakes the jarring edge off of loud sounds but allows the quiet sounds(leaves rustling etc.) to maintain their original level. This is a goodmode for loud volume settings. Thus, a fuller dynamic range can beachieved while still compressing loud annoying signals. Anotherarrangement involves heavy compression (for example 10:1) for AVG valuesabove and below the Level threshold. Heavy compression is referred toherein as a “night mode” since you can hear all sounds in the program(both loud and soft) without having to turn the volume up (for softsounds) and down (for loud sounds). Night mode is good for low volumesettings, which are often preferred by television viewers during thelate night hours.

Even further, another embodiment contemplates the use of high and lowattack and release ratio thresholds. In such an embodiment, the twothresholds define three regions of a loudness space: quiet, normal, andloud. In each of these windows, a different compression ratio can beapplied. For example, a 1.5:1 ratio can be used to amplify quietsignals, a 1:1 ratio can be used to preserve normal signals, and a 4:1ratio can be used to attenuate loud signals. With this multi-windowedsystem, the original dynamic range can more accurately be preservedwhile fringe loud and soft signals can be attenuated and amplifiedrespectively.

Lastly, if the processing is performed in the log domain, then thecalculated compression ratio is “linearized” at 140 before applying thegain to the input signal.

FIG. 3 shows a single band system 300 wherein one DVC system 302 canapply the same gain to each of the left (L) and right (R) signalsapplied to the respective inputs 304 and 306. Specifically, as seen inFIG. 3, the output of the DVC system 302 (provided by the log-to-linearsignal converter 140) dynamically sets the gain of each of theamplifiers 308 and 310 respectively, which in turn amplify thecorresponding left and rights signals applied to the two inputs of thesystem 300 providing the Lout and Rout signal at the outputs 316 and318. The DVC system 302 can be responsive to the entire frequency rangeof each of the L and R signals, or only a selective band of each asshown in FIG. 3 for example, high pass filters 312 and 314 each onlypass a high frequency portion of the respective L and R signals to theDVC system 302, so that the latter only responds to high frequencycontent of each of the signals.

Alternatively, a multi-band system can be configured so that selectbands are each individually processed by its own DVC system so the L andR signals are independently controlled. As shown in FIG. 4, for example,a two band system 400 employs two DVC systems 406 and 408, each for theL and R signals, so that that L and R signals applied to the inputs 402and 404 enjoy independent gain control. As shown, the L signal isapplied to a high pass filter 410 and low pass filter 412, while the Rsignal is applied to the high pass filter 414 and low pass filter 416.In a two band system of FIG. 4 with high and low bands, a DVC system(406 and 408) can apply a gain to the L and R signals in the high bandby applying the output of each DVC system to the respective outputs ofthe high and low pass filters. Specifically, the output of DVC system406 is applied to control the gain of each of the amplifiers 418 and 420which receive and amplify the high frequency outputs of high passfilters 410 and 412. Similarly, the output of DVC system 408 is appliedto control the gain of each of the amplifiers 422 and 424 which receiveand amplify the low frequency outputs of the low pass filters 412 and416. The outputs of amplifiers 418 and 420 are added at signal summer426 so as to produce the output signal Lout at output 428, while theoutputs of amplifiers 422 and 424 are added by the signal summer 430 soas to produce the output signal Rout at output 432.

In another embodiment, if independent gain control of each L and Rsignal in a multi-band signal is desired, then a separate DVC system canbe used for each band of each of the L and R signals. Further, insteadof a multi-band system, a high pass filter can be used to eliminate lowfrequencies for systems unresponsive to low frequencies such as shown inFIG. 3.

Regarding the filters used with the multi-band DVC system, the crossover frequency between each contiguous band (in the two band system thiswould be the low and high pass bands) can be adjustable. It is alsopossible to leave the cross over frequency fixed. One example is acrossover based upon a digital implementation of a derived filter.Derived filters are described in THAT Corporation Application Note 104from THAT Corporation of Milford, Mass., and in Bohn, D. (Ed.), AudioHandbook (National Semiconductor Corporation, Santa Clara, Calif. 1976)§5.2.4. In one example of a derived filter implementation, the crossoveruses a 2^(nd) order Butterworth LPF and a derived HPF which sum to unityas shown in FIG. 5. In another example, the crossover is a traditionaldigital 2nd order with a Q=0.5 with the HPF inverted so the bands sum tounity as shown in FIG. 6. In yet another example, the crossover is basedon 4th order Linkwitz-Riley filters which sum to unity as shown in FIG.7. In the single band volume control a high pass filter controls theinput of the RMS detector.

Multi-Spatial Processing Protection (MPP)

Television manufacturers often include virtual surround (pseudosurround)technology (e.g., SRS Tru-Surround, Spatializer etc.) in the two-channeltelevision audio output path. This two-channel television audio may goto speakers external to the television or to speakers mounted in thetelevision enclosure. These virtual surround technologies create theillusion of surround sound by manipulating and enhancing the differencechannel (L−R) present in stereo broadcasts. The listener still perceivesan intact center image (L+R) but also often hears the difference channel(L−R) either widened over a broad soundstage or as a point sourcelocated somewhere other than the speaker locations. Often this type ofspatial enhancement is done during the production of the audioprogramming. This is especially true of television commercials which areenhanced to grab the listener's attention. When an audio program has twocascaded stages of spatial enhancement (for example at the point ofproduction and in a television's audio processing) there can besignificant degradation in the audio quality. The preprocessed audiotends to have significant L−R energy relative to L+R energy. The second,cascaded stage, of spatial enhancement processing tends to increase theamount of L−R energy even more. Recent studies have shown that excessiveamounts of L−R enhancement is one of the top factors in listenerfatigue. There also can be a significant volume increase.

Accordingly, in accordance with one aspect of the invention, a MPPsystem is provided. In one embodiment the MPP is a double processingprotection (DPP) system that is a part of a television audio signalreception and playback system, prior to the television's stereoenhancement technology. The MPP system is hereinafter referred to as apseudosurround signal processor. The exemplary DPP system processes theaudio signals so as to minimize the difference (L−R) enhancement (i.e.,minimizing the energy level of the difference (L−R) signal relative tothe sum (L+R) signal) introduced at the point of production. This allowsthe television's spatial enhancement technology to process the audiosignals in a manner that is psychoacoustically pleasing to the listener.The cascade of the DPP system before the television's spatialenhancement audio processing has proven to be quite effective inmitigating the harsh effects of double spatial processing. In oneembodiment the DPP system is entirely digital and can be implementedeconomically in software (C, assembler etc.) or digital hardware (HDLdescription). It should be appreciated that the DPP system can also beall analog, or a hybrid of analog and digital components.

In one embodiment the DPP system reduces L−R enhancement relative to thecorresponding L+R level. The embodiment reduces the effects of multiple2 channel spatial effects processing. One embodiment of such a system isshown in FIG. 8 at 800. The left signal L and the right signal R arerespectively applied to the inputs 802 and 804 of system 800. The L andR signals are applied to matrices represented by the two signal summers806 and 808. Signal summers 806 and 808 constitute the matrix whichprovides the SUM (L+R) and DIF (L−R) signals.

In the sum (L+R) path, the signal is generally untouched. The SUM signalusually contains audio content which does not necessarily need to belocalized. However, in alternate embodiments, frequency contour shapingcan be performed to enhance audio content such as dialogue. As shown,the SUM signal is multiplied by a Center constant at signal multiplier810 prior to be provided to matrices illustrated as signal summers 812and 814. The Center constant allows the level of the center image (L+R)to be adjusted, if desired, to aid in intelligibility of dialogue.Adding the L+R and L−R signals provides the left output signal Lo atoutput 816, while subtracting the L−R from the L+R provides the rightoutput signal Ro at output 818.

In the illustrated embodiment of FIG. 8, most of the processing occursin the DIF path. L+R and L−R are compared to determine the level of theL−R signal relative to L+R. Before comparison, these two SUM and DIFsignals can be each passed through a respective high pass filter 820 and822, such as in circumstances where the speaker frequency response doesnot include low frequencies. The L−R DIF signal can further be passedthrough a multi-band equalizer 824 to accentuate the ear's mostsensitive frequencies, namely mid-range frequencies, to compensate forthe perceived loudness level of the L−R signal. Equalizer 824 allows thedifference channel level detection to be frequency dependent. Forexample, low frequency signals may be minimized when processing forinexpensive television speakers with limited bass response. Highfrequencies may be minimized to limit the response to transient audioevents. Typically mid range frequencies, were the ear is most sensitive,are equalized to dominate the difference level detection. Once thelevels of the difference and sum signals are calculated the DIF/SUMratio is determined.

Each of these signals is then run through a respective signal leveldetector 828 and 830. The detectors listed above can be used, such as anRMS level detector, although any type of level detector (such as theones described above) can be used. Also, the processing can all beperformed in the log domain to increase efficiency by processing themthrough the log domain processing blocks 832 and 834.

The outputs of the blocks 832 and 834 are applied to the signal summerwherein the processed SUM signal is subtracted from the processed DIFsignal. Subtracting one signal from the other in the log domain is thesame as providing a signal that is the ratio of the process SUM signalto that of the DIF signal in the linear domain. Once the L+R and L−Rsignal levels are calculated, where the L−R signal level may have beenequalized prior to level detection to increase the mid-rangefrequencies, these two signal levels are compared by the comparator 838to a preset threshold 840. The ratio between the two signals((L−R)/(L+R)) is compared to a threshold ratio by comparator 838 inorder to determine the recommended L−R signal gain adjustment. A limiterstage 842 may be used to limit the amount and direction of gain appliedto the L−R signal. The illustrated embodiment limits the gain at 0 dBhence only allowing attenuation of the L−R signal, although in someapplications, there may be a desire to amplify the L−R signal. Anaveraging stage 844 averages, with a relatively long time constant, theoutput of the limiter stage 842 so as to prevent the DPP system fromtracking brief transient audio events. After conversion back to thelinear domain by linear domain block 846, the level of the L−R signal iscorrespondingly adjusted by the signal multiplier 848 to achieve thattarget ratio.

Even in the absence of multiple stages of spatial preprocessing thetarget (L−R)/(L+R) ratio can be set low to allow, for example, anincreased intelligibility of program dialogue.

Another method and system for double processing protection is to“predict” the preprocessing performed on the L−R signal and compensatefor the preprocessing from the prediction. For example, if SRSTru-Surround is known to be used on L−R, then the signal cancorrespondingly be compensated to remove the L−R enhancement.Alternatively, the signal energy can be monitored over time to deducethe pre-processing performed on the L−R signal. From this deduction, theL−R signal can be compensated to remove any such L−R enhancements.Preprocessing could change the frequency response of the difference (andsum for that matter) channel as well as the L−R/L+R ratio. The inversefilter, of the preprocessor, could be applied to each path while theexisting L−R/L+R ratio adjustment still remains in use.

Further, while the DPP system of FIG. 8 is shown as a feed forwardsystem wherein the DIF signal is sensed prior to the variable gaincontrol amplifier 848, a feedback system, wherein the sum and differencesignal levels are detected after the variable gain control amplifier isalso possible.

Combining DVC and DPP

Since each of the DVC and MPP provide an improved listening experience,the two can be combined to combine the advantages of both. There are anumber of ways of combining DVC and DPP blocks. One example of a usefultopology places the DPP block 902 first, followed by a DVC block 904 ina cascaded design, as shown in FIG. 9. In this embodiment, the L and Rsignals are applied to the inputs 906 and 908 of the DPP block 902. TheL′ and R′ outputs of the DDP block 902 at outputs 910 and 912 areapplied to the two inputs 914 and 916 of the DVC block 904. The outputs918 and 920 of DVC block provide the respective output signals Lo andRo. The cascaded design allows the DPP block to remove the difference(L−R) signal enhancement first, then maintain the perceived constantlevel of the stereo audio program with the DVC block without ambientenergy being present.

Another example of a topology places the DPP block 1004 in a feedbackpath of the DVC block 1002, as shown in FIG. 10. The L and R inputs areapplied to the inputs 1006 and 1008, respectively. The two signals areapplied to matrices (represented by signal summers 1010 and 1012) so asto produce the SUM (L+R) signal and the DIF (L−R) signal. The outputs1014 and 1016 of the DVC block 1002 provide the outputs signals Lo andRo. The two outputs 1014 and 1016 provide the two feedback signals ofthe feedback path. Specifically, the Lo and Ro signals are applied tomatrices shown as to signal summers 1018 and 1020 so that the Lo+Roforms one input of the DPP block 1004, and the Lo−Ro forms the otherinput of the DPP block 1004. The output of the DPP block 1004 representsthe corrected gain, which is then applied to the DIF signal by signalmultiplier 1022. The latter can be in the form of a variable gaincontrol amplifier. It should be appreciated that while two embodimentsof the combined DVC and DPP blocks are illustrated in FIGS. 9 and 10,other combinations are possible.

Accordingly, embodiments of the present disclosure can provide forimproved performance of audio signal reproduction which reduces theeffects of undesirable volume changes in audio programming.

The components, steps, features, benefits and advantages that have beendiscussed are merely illustrative. None of them, nor the discussionsrelating to them, are intended to limit the scope of protection in anyway. Numerous other embodiments are also contemplated. Additionally,embodiments of the present disclosure can have fewer, additional, and/ordifferent components, steps, features, benefits and advantages than asexpressly described herein. These also include embodiments in which thecomponents and/or steps are arranged and/or ordered differently.

Unless otherwise stated, all measurements, values, ratings, positions,magnitudes, sizes, and other specifications that are set forth in thisspecification, including in the claims that follow, are approximate, notexact. They are intended to have a reasonable range that is consistentwith the functions to which they relate and with what is customary inthe art to which they pertain.

All articles, patents, patent applications, and other publications whichhave been cited in this disclosure are hereby incorporated herein byreference.

The phrase “means for” if and when used in a claim is intended to andshould be interpreted to embrace the corresponding structures andmaterials that have been described and their equivalents. Similarly, thephrase “step for” if and when used in a claim embraces the correspondingacts that have been described and their equivalents. The absence ofthese phrases means that the claim is not intended to and should not beinterpreted to be limited to any of the corresponding structures,materials, or acts or to their equivalents.

Nothing that has been stated or illustrated is intended or should beinterpreted to cause a dedication of any component, step, feature,object, benefit, advantage, or equivalent to the public, regardless ofwhether it is recited in the claims.

The scope of protection is limited solely by the claims that now follow.That scope is intended and should be interpreted to be as broad as isconsistent with the ordinary meaning of the language that is used in theclaims when interpreted in light of this specification and theprosecution history that follows and to encompass all structural andfunctional equivalents.

1. A system for dynamically controlling the perceived volume of a stereoaudio program including left and right channel signals, comprising: adynamic volume control configured and arranged so as to maintain aperceived constant volume level of the stereo audio program; and aprogram change detector configured and arranged to provide a programchange signal indicating that the volume of the left and right channelsignals has dropped below a threshold level for at least a thresholdtime period so as to anticipate a possible change in the sound level ofthe left and right channel signals; wherein the dynamic volume controlis responsive to the program change signal.
 2. A system according toclaim 1, wherein the dynamic volume control includes a compressorconfigured and arranged to have adjustable speeds of an attack time anda release time, wherein if a program change signal is detected thespeeds of the attack and release times are increased.
 3. A systemaccording to claim 2, wherein the dynamic volume control is configuredand arranged so that if a loud channel is changed to a quiet channel,then the increased release time permits a faster gain increase to meet atarget sound output level, and if a quiet channel is changed to a loudchannel, then the increased attack time permits a faster gain decreaseto meet the target sound output level.
 4. A system according to claim 3,wherein the dynamic volume control is configured and arranged so that ifthe sound level rises above the threshold level before the time expires,then a program change is not detected.
 5. A system according to claim 1,wherein the threshold time period and the threshold level are fixed. 6.A system according to claim 1, wherein the threshold time period and thethreshold level are adjustable.
 7. A system according to claim 1,wherein the threshold time period and the threshold level are variable.8. A system according to claim 1, wherein the program change detector isconfigured and arranged so as to respond automatically by detecting amute condition during a channel change.
 9. A system according to claim1, wherein the program change detector is configured and arranged so asto respond by detecting a channel change condition from a user's remotecontrol when the user is changing a channel.
 10. A system according toclaim 1, wherein the program change detector is configured and arrangedso as to respond to a channel change condition communicated via a hostprocessor.
 11. A method of dynamically controlling the perceived volumeof a stereo audio program including left and right channel signals,comprising: dynamically controlling the volume of the left and rightchannel signals in response to a program change signal so as to maintaina perceived constant volume level of the stereo audio program; andgenerating the program change signal in response to detecting a programchange signal indicating that the volume of the left and right channelsignals has dropped below a threshold level for at least a thresholdtime period so as to anticipate a possible change in the sound level ofthe left and right channel signals.
 12. A system for dynamicallycontrolling the perceived volume of a stereo audio program includingleft and right channel signals, comprising: a dynamic volume controlconfigured and arranged so as to maintain a perceived constant volumelevel of the stereo audio program, the dynamic volume control includingat least one compressor responsive to high and low attack and releaseratio thresholds so as define quiet, normal and loud perceived volumelevels.
 13. A system according to claim 12, wherein the compressorapplies a different compression ratio to each of the quiet, normal andloud perceived volume levels.
 14. A system according to claim 13,wherein the compression ratio for quiet perceived volume levels is setto amplify the left and right signals, the compression ratio for normalvolume levels is set to preserve the left and right signals, and thecompression ratio for loud volume levels is set to attenuate the leftand right signals.